similar to: Asterisk, Configuration of SDP in SIP messages

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the

I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the same setup, the calling party is unable to hear the IVR recording from the asterisk installation, although in fact the streaming is supposed to

Hi, I am using codec g729 on two asterisk machines, but when call is forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs following error and there is no audio. Also the IVRs being played have choppy voice. "Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = '')" It is running fine when codec gsm is in RTP traffic. Also I

Hello, If I understand it correctly, when sending INVITE, o= and c= sdp fields are built using p->ourip IP address. At this point RTP packets will be coming to the default asterisk IP address. For the machine with multiple interfaces this could be not the right one (not what we want). Could it be configured (in rtp.conf or in sip.conf per context) ? Thank you. Alex Zarubin --------------

Hello list when trying to set up webRTC communications with sipjs client package (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file the following : DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 99.88.77.66... OK. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED.

Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called

Hello, Is there a way to configure Asterisk so that it doesn't care about the playload type in SDP ? I'm trying to send custom data which has assigned a dynamic PT through RTP so I only need Asterisk to act as a proxy/forwarder but I'm getting "488 Not acceptable here" responses and in console "No compatible codecs, not accepting this offer" Best regards, Nicolae

Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8

Hi Is it a normal behavior of Asterisk put a call on hold when receive a Session Progress with media address 0.0.0.0 in SDP? I believe the call on hold should be initiate with a re-invite. Thanks -- Att, Rafael Saraiva -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this

Hi everyone, samba 2.2.5 The server I'm using has 2 interfaces so using the interface parameter I'm telling samba to use eth0 but for some reason when I do netstat it is listening on eth1 interface = eth0 (the IP is 192.168.6.10) netstat -an udp 0 0 138.79.161.225:137 0.0.0.0:* udp 0 0 0.0.0.0:137 0.0.0.0:* udp 0 0 138.79.161.225:138

Hello, The setup is : Asterisk ------<www> ------- Router1---------<LAN> --------- SIP Phones | | |------------------<pstn>----------Router2------------| (I hope the sketch is understandable. If not, it could be summarized with : "SIP hardphones on a given LAN are connected to an Asterisk server through 2

Hello, I was testing with sdp and something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine,

Hi Guys, Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22. I have a provider who re-invites with the following sdp (message flow PROVIDER_EQPMT -> ASTERISK): """ . v=0. o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. s=-. c=IN IP4 CONN_IP_PROVIDER. t=0 0. m=audio 0 RTP/AVP 0. m=image 26858 udptl t38. a=T38FaxMaxBuffer:288.

I have an FC5 system with a powerware 5110 usb ups. The system seems to find the UPS ------------------------------ [root@b1 src]# lsusb Bus 004 Device 001: ID 0000:0000 Bus 003 Device 002: ID 0592:0002 Powerware Corp. Bus 003 Device 001: ID 0000:0000 Bus 002 Device 001: ID 0000:0000 Bus 005 Device 001: ID 0000:0000 Bus 001 Device 001: ID 0000:0000 ------------------------------- [root@b1

Dear all, I have Asterisk 1.2.13 running OK with Twinkle clients, they can talk very well using SIP. I have a Jabber server running OK and the clients use PSI client for chat succesfully. Now I'm using Wengophone 2.1.1 in order to unify voip+IM services. The users can logon OK in SIP and Jabber, they get the online status presence, but they CAN'T talk and chat among them.

We're dialing a disconnected number via Level 3's vector network, and are receiving this. The response has SDP in it. Apparently, Level 3 is playing early media. Asterisk doesn't seem to know what to do with SDP in a 180 RINGING, and just plays ringing. What am I missing here? How can Asterisk see there's SDP, early media, in the response and act accordingly? SIP/2.0 180

Hello: I have this situation: I can make calls internally, I can make inbound calls but I can't make outbound calls. Thanks in advance. These are my devices: * asterisk 11.8.1 = 192.168.1.22 * sipphone grandstream gxp2160 = 192.168.1.5 * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4 port 1 (FXS) connected to an analog phone port 3 (FXO) connected to the PSTN These are my